asterisk users - May 2010 - Re-invite from Asterisk Server: Port number changes

Hello list,

I am trying to test a scenario wherein two clients configured on two diffrent
boxes try to communicate with each other by means of Asterisk. The softphone on
both the boxes is zoiper. One of the boxes is Unix, and has the server running
on it. The other is Windows.

When I make a call between clients (Unix -> Windows), the signaling works
fine, but I cannot listen to audio on the Windows box. The audio device on the
Windows box has been tested to be working fine.

Upon analyzing the wireshark SIP logs, I can see that Asterisk sends a re-invite
in which the RTP port number in SDP is different from that configured on zoiper.

I suspect that this is probably the reason for the audio non-availability on the
Windows box. I have enabled allowdirectmedia as part of my SIP settings.

I have posted the logs at clipnet:

http://cl1p.net/sip.conf (SIP.conf used for this test)
http://cl1p.net/Call_logs_Unix_lo (Wireshark logs captured on the local
interface on the Unix box)
http://cl1p.net/Call_logs_Unix_eth0 (Wireshark logs captured on the eth0
interface on the Unix box)
http://cl1p.net/Call_logs_Windows (Wireshark logs captured on the Windows box)

I would appreciate your thoughts.

Thanks
Vin